- FreeSWITCH-users - FreeSWITCH Mailing Lists

Can't make mod_rtmp
by peely 07 Sep '11

07 Sep '11

Hi, I'm having trouble building FreeSWITCH with the latest mod_rtmp goodness. On Ubuntu X64 compiling with gcc I get: cc1: warnings being treated as errors /usr/src/freeswitch/src/mod/endpoints/mod_rtmp/mod_rtmp.c: In function ‘rtmp_function’: /usr/src/freeswitch/src/mod/endpoints/mod_rtmp/mod_rtmp.c:1612:114: error: comparison between ‘switch_call_cause_t’ and ‘enum <anonymous>’ I'm not sure if I need to do anything else, I just naively added endpoints/mod_rtmp to modules.conf and did a make clean install? Thanks, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Can-t-make-mod-rtmp-tp6488229… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Is there a SIP client that is capable of displaying arbitrary SIP headers?
by Yungwei Chen 07 Sep '11

07 Sep '11

Hi, I'm wondering if there's a SIP client that can display arbibrary SIP headers on its UI. Thanks.

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Overlap dialing and not presented 'sending complete' flag
by Štefan Čudai 07 Sep '11

07 Sep '11

Dear Friends, my questions are about FreeTDM and his module sangoma_isdn. I have to use overlap dialing because of customer requirements. First question: In file ftmod_sangoma_isdn_stack_hndl.c on line [474] is following condition: if (cnStEvnt->sndCmplt.eh.pres || num_digits >= min_digits) { ftdm_set_state(ftdmchan, FTDM_CHANNEL_STATE_RING); } else { ftdm_log_chan(ftdmchan, FTDM_LOG_DEBUG, "received %d of %d digits\n", num_digits, min_digits); } I guess it should be && and no || because when you have set min_digits = 8 channel goes immediately to RING state after INFO message with eighth digit is received. Than it isn't possible receive numbers longer then 8 digits! Is it BUG??? Second question: According to "ITU-T Recommendation Q.931" in chapter "5.1.3 Overlap sending" is written: --------------------------------------------------------------------------------------------------- The call information in the message which completes the information sending may contain a sending complete indication, (e.g. the # character or, as a network option, the sending complete information element) appropriate to the dialling plan being used. The network shall restart timer T302 on the receipt of every INFORMATION message not containing a sending complete indication. -------------------------------------------------------------------------------------------------- If I understand this flag "cnStEvnt->sndCmplt.eh.pres" mentioned in the code above is used to find out if INFO message has 'sending complete indication'. But what about situation when FreeTDM doesn't get it??? In that case FreeTDM received after some time (in my case 15s) STATUS CONFIRM message with reason: Recovery on timer expire (call_state:25 channel-state:COLLECT cause:102) (suId:1 suInstId:2 spInstId:2) Can you tell me if implementation of sangoma_isdn module also solves this situation when 'sending complete' is not set? Thanks. Regards, Stefan

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[mod_xml_curl] Keep receiving 403 forbidden
by e1s 07 Sep '11

07 Sep '11

I keep on receiving 403 Forbidden when try to register using mod_curl_xml, here's the log lines : http://pastebin.freeswitch.org/17293 my curl return : <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="directory" description="User Directory"> <domain name="10.1.1.47">" <user id="6597293279"> <params> </params> </user> </domain> </section> </document> I couldn't find what's wrong.. ----- /e1.s -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-xml-curl-Keep-receiving-4… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Caller-Privacy-Hide-Name: [true]
by Luis F Urrea 07 Sep '11

07 Sep '11

Hi all, Any one can shed some light on why this channel variable Caller-Privacy-Hide-Name: [true] suddenly got set to true for a single phone ? Is this something that can be set by the phone through the SIP INVITE? or maybe in the directory? This got turned on on a Snom 370 which is the receptionist phone and threfore all calls from this phone appear as anonymous. Nothing was changed on the dial plan or even directory. This is a way old build of FS just FYI. Your help is greatly appreciated. Regards, Luis

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Opposite of flush_dtmf - grab dtmf from before the IVR + debug rfc2833
by Avi Marcus 06 Sep '11

06 Sep '11

I have a calling card IVR and upon an error, it announces the error, then goes back to the calling card IVR. How can I use the digits they started dialing during the pre-IVR announcement as part of the IVR? e.g. http://pastebin.freeswitch.org/17283 During the announcement of the "wrong" dialed number, I started dialing and FS got the digits. But it didn't start "listening" until... just before (??) the calling_card_short IVR started. How can I get it to listen to everything that got queued up? Thanks, -Avi p.s. is there somelinke like ngrep/tcpdump on regular SIP packets to live-view the rfc-2833 that come in, before they even hit FS? I know wireshark can understand them in a pcap, but I couldn't figure out how to filter live...

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bridge after IVR delayed/no media in git-2689081 2011-09-01 ?
by Avi Marcus 06 Sep '11

06 Sep '11

Soon after updating to GIT FreeSWITCH Version 1.0.head (git-2689081 2011-09-01 21-16-22 -0500) I started getting some complaints on my calling card that they wouldn't hear anything. I see the calls in my CDR are marked as the other party picking up, across several carriers. I have a recording where after the bridge, there seems to be a long delay on the A leg media to start to show up in the recording, and A couldn't hear B, either (who was in the recording). I WOULD upgrade to the latest GIT and test, but it doesn't happen all the time. In fact, since I turned on call recording, this delay is the only issue I've seen yet. Did something change around that date that might have caused this? I did several normal SIP calls (no IVR involved) and didn't have any issues with hearing the B leg... -Avi

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External/internal profile
by Anto 06 Sep '11

06 Sep '11

Hello Today I can not understand the internal and external profiles. I've used other times asterisk setup and both the carriers and customers share the same ports and same ips. I do not quite understand because FreeSWITCH must use different ports (do not know if you can configure the same ports. In all the how-to I've seen that change the profiles and use different ports). Could anyone guide me for this?. If for example I have a server with two public IPs, you could configure each profile with an ip and both with the same ports (5060 and 5061)?. Perhaps it is an obvious question, but do not quite understand the issue of profiling, I have doubts. Thank you very much. Best regards.

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Will bypass-media work when a call is bridged to multiple SIP destination
by dma 06 Sep '11

06 Sep '11

Hi Support, This is a simple question but I don't find an answer from the web. I hope to know whether bypass-media work when a call is bridged to multiple SIP destination? Thanks, D.Ma -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Will-bypass-media-work-when-a… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Variables in Dialplan - Problem with getting Variables from User Directory
by Silvio Escher 06 Sep '11

06 Sep '11

Hi there, iam really getting crazy these days ;-) I've changed our Freeswitch Version some Weeks ago ( cannot remember exactly but proably from 1.0.6 ) to the git Head. ). Since this Change ( or better since the followed Config File Adaptions ;-) ) i noticed that i cannot use User Variables/Params inside the XML-Dialplan anymore. Some (relevant?! ;) ) Copy&Paste .. my Userentry <include> <user id="99"> <params> <param name="password" value="something"/> <param name="vm-mailto" value="maybe_me(a)omeco.de"/> <param name="vm-mailfrom" value="noreply(a)omeco.de"/> <param name="vm-enabled" value="true"/> <param name="vm-password" value="user-choose"/> <param name="vm-email-all-messages" value="false"/> <param name="vm-notify-email-all-messages" value="true"/> <param name="vm-keep-local-after-email" value="true"/> </params> <variables> <variable name="outbound_caller_id_name" value="Omeco GmbH"/> <variable name="outbound_caller_id_number" value="06316803540"/> <variable name="effective_caller_id_name" value="Silvio Escher"/> <variable name="effective_caller_id_number" value="40"/> <variable name="callgroup" value="1"/> </variables> </user> </include> some dp snippet.. before the change something like <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> was working fine .. Actually iam using <action application="set" data="effective_caller_id_name=${user_data(${username}@${domain_name} var outbound_caller_id_name)}"/> <action application="set" data="effective_caller_id_number=${user_data(${username}@${domain_name} var outbound_caller_id_number)}"/> or an <action application="set_user" data="$1@${domain_name}"/> to get some or all Variables. But i Noticed that the "set_user" Thingie just gets me the Variables - not the Params. So i've still Issues with mod_voicemail ( Mailto is undef ) 2011-08-30 22:02:32.904211 [DEBUG] switch_utils.c:709 Emailed data to [(null)] 2011-08-30 22:02:32.904211 [DEBUG] mod_voicemail.c:2809 Sending notify message to (null) Maybe i has something todo with an missing auth or wrong Domain or whatever .. but i have no clue where to look further. Any Help ( from little Hints till complete Miracle Solve ;-) ) is actually very welcome. Thanks in Advance, Silvio -- Silvio Escher omeco GmbH

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